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<p class="MsoNormal"><span style="color:#1F497D">Any solution to this. Any help will be appreciated.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Thanks<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D">Vikas<o:p></o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><span style="color:#1F497D"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span lang="EN-US" style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Vikas Kumar<br>
<b>Sent:</b> 22 September 2015 21:35<br>
<b>To:</b> freeswitch-users@lists.freeswitch.org<br>
<b>Subject:</b> [Freeswitch-users] Audio files are not bein played<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hi All, <o:p></o:p></p>
<p class="MsoNormal">We are using Linphone and zoiper on other hand. We have some limit in dial plan that only one call is admissible at one time.<o:p></o:p></p>
<p class="MsoNormal">If I call from other softphone while call is in progress than this limit is hit. See logs:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Dialplan: <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Action playback(/usr/local/videocallhandler/media/InQue1.wav)<o:p></o:p></p>
<p class="MsoNormal">Dialplan: <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Action playback(/usr/local/videocallhandler/media/id.wav)<o:p></o:p></p>
<p class="MsoNormal">Dialplan: <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Action playback(/usr/local/videocallhandler/media/InQue2.wav)<o:p></o:p></p>
<p class="MsoNormal">Dialplan: <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Action endless_playback(/tmp/times.wav)<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:167 (<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>) State Change CS_ROUTING -> CS_EXECUTE<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_session.c:1357 Send signal
<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a> [BREAK]<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>) State ROUTING going to sleep<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:418 (<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>) Running State Change CS_EXECUTE<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:481 (<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>) State EXECUTE<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> SOFIA EXECUTE<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:209
<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a> Standard EXECUTE<o:p></o:p></p>
<p class="MsoNormal">EXECUTE <a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a> answer()<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000]<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3216 Set Codec
<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a> GSM/8000 20 ms 160 samples 13200 bits<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_core_codec.c:111 <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Original read codec set to GSM:3<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5477 Set 2833 dtmf send/recv payload to 101<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3475 AUDIO RTP [sofia/internal/1016@119.9.73.183] 119.9.73.183 port 25094 -> 10.150.178.79 port 60204 codec: 3 ms: 20<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_rtp.c:2041 Starting timer [soft] 160 bytes per 20ms<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3742 Set 2833 dtmf send payload to 101<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3748 Set 2833 dtmf receive payload to 101<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3775 <a href="mailto:sofia/internal/1016@119.9.73.183">
sofia/internal/1016@119.9.73.183</a> Set rtp dtmf delay to 40<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [NOTICE] sofia_glue.c:4386 Pre-Answer
<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>!<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>) Callstate Change RINGING -> EARLY<o:p></o:p></p>
<p class="MsoNormal">2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:866 Local SDP
<a href="mailto:sofia/internal/1016@119.9.73.183">sofia/internal/1016@119.9.73.183</a>:<o:p></o:p></p>
<p class="MsoNormal">v=0<o:p></o:p></p>
<p class="MsoNormal">o=FreeSWITCH 1442877029 1442877030 IN IP4 119.9.73.183<o:p></o:p></p>
<p class="MsoNormal">s=FreeSWITCH<o:p></o:p></p>
<p class="MsoNormal">c=IN IP4 119.9.73.183<o:p></o:p></p>
<p class="MsoNormal">t=0 0<o:p></o:p></p>
<p class="MsoNormal">m=audio 25094 RTP/AVP 3 101<o:p></o:p></p>
<p class="MsoNormal">a=rtpmap:3 GSM/8000<o:p></o:p></p>
<p class="MsoNormal">a=rtpmap:101 telephone-event/8000<o:p></o:p></p>
<p class="MsoNormal">a=fmtp:101 0-16<o:p></o:p></p>
<p class="MsoNormal">a=ptime:20<o:p></o:p></p>
<p class="MsoNormal">a=sendrecv<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Warning given is “/usr/local/videocallhandler/media/Hold.wav sample rate 11025 doesn't match requested rate 8000”<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Should I ignore this warning, ??<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks<o:p></o:p></p>
<p class="MsoNormal">Vikas<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<table><tr><td bgcolor=#ffffff><font color=#000000><pre>Disclaimer :-
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be those of the specific TIMES GROUP company.</pre></font></td></tr></table>